VoIP Telephone Adapter User s Manual

VoIP Telephone Adapter User s Manual Last Update: 2008/10/10

1 Introduction...3 1.1 Product Overview (Single Phone Port Model)...3 1.2 Product Overview (Dual Phone Port Model)...4 2 IVR Interface for TA...6 3 Setup the TA by Web Browser...7 3.1 First Time Login...7 3.2 Default Reset from Keypads...9 3.3 Default Setting...9 4 Application Example...9 4.1 SIP-to-SIP Calling/Answering...9 4.2 SIP to Direct IP Calling...9 4.3 Direct IP to Direct IP Calling/Answering... 10 4.4 Direct IP to Direct IP Calling within NAT Router... 10 4.5 Calling through the FXO Port (LINE Port)... 10 4.6 3-Way Conference Call, Call Waiting, Call Hold... 11 4.6.1 3-Way Conference Calling Application... 11 4.6.2 Call Waiting Application:... 11 4.6.3 Call Hold Application:... 11 4.6.4 Call Transfer Blind Transfer... 11 4.6.5 Call Transfer Attendant Transfer... 11 4.6.6 Call Forward... 11 5 Phone Configuration... 11 5.1 System Information... 11 5.2 Phone Book... 12 5.3 Phone Setting... 13 5.3.1 Call Forward function... 13 5.3.2 SNTP setting function... 15 5.3.3 Volume setting function... 15 5.3.4 DND function... 16 5.3.5 Auto Answer function (Supported by FXO Model Only)... 16 5.3.6 Trunk Gateway (Supported by FXO Model Only)... 17 5.3.7 Caller ID function... 18 5.3.8 Flash Time Setting function... 18 5.3.9 Call Waiting setting function... 19 5.3.10 T.38 Fax setting function... 19 5.3.11 Dial Plan... 19 5.3.12 Dial Plan Call Routing (Supported by FXO Model Only)... 21 5.4 Network... 21 5.4.1 Network Status... 21 5.4.2 WAN Settings... 22 5.4.3 LAN Settings... 23 5.4.4 DDNS Setting... 23 5.4.5 VLAN Setting... 24 5.4.6 PPTP Setting... 24 5.4.7 STUN Setting... 25 5.5 SIP Settings... 25 5.5.1 Service Domain... 25 5.5.2 Port Settings... 26 5.5.3 Codec Settings... 26 5.5.4 Codec ID Setting... 27 5.5.5 DTMF Setting... 28 5.5.6 RPort Function:... 28 5.5.7 Other Settings... 28 5.6 Advanced... 29 5.6.1 Auto Configuration Setting... 29 5.6.2 FXS/FXO Setting:... 30 5.6.3 Advanced Setting... 31 5.7 System Auth... 32 5.8 Save Change... 32 5.9 Update... 32 5.10 Reboot... 33 6 Troubleshooting... 34

1 Introduction This user s manual is for VoIP Telephone Adapter (TA). This user s manual will explain the IVR instruction, web configuration and command line configuration for the TA. Before using the TA, some setup processes are required to make the TA work properly. Please refer to the Setup Menu for further information. 1.1 Product Overview (Single Phone Port Model) The TA has the following interfaces for Networking, telephone interface, LED indication, and power connector. Networking These two interfaces support 10/100Mps Fast Ethernet. WAN: Connect to the ADSL or Router. LAN: Connect to your PC computer. You can then connect to Internet through the Internet Sharing function of the Adapter. Phone PHONE: Connect to the standard (analog) phone or Fax machine LINE: (Optional) Connect to the landline (PSTN), usually presented as a phone jack on the wall of your home or office LED Light Indicators There are four LED indicators in the TA to show the Power, WAN, LINE and VOIP indication. Power: On when Power is on WAN: On when the WAN network link is up LINE: On when the phone is switched to regular landline (PSTN). VOIP: On when VoIP service is registered, blink when the VoIP call is active.

1.2 Product Overview (Dual Phone Port Model) The TA has the following interfaces for Networking, telephone interface, LED indication, and power connector. Networking These two interfaces support 10/100Mps Fast Ethernet. WAN: Connect to the ADSL or Router. LAN: Connect to your PC computer. You can then connect to Internet through the Internet Sharing function of the Adapter. Phone PHONE1: Connect to a standard (analog) phone or Fax Machine PHONE2: Connect to second standard (analog) phone or Fax Machine LED Light Indicators There are four LED indicators in the TA to show the Power, WAN, PHONE1 and PHONE2 indication. Power: On when Power is on WAN: On when the WAN network link is up PHONE1: On when VoIP service of Phone 1 is registered, blink when Phone 1 is active. PHONE2: On when VoIP service of Phone 2 is registered, blink when Phone 2 is active.

1.1 Software Overview Network Protocol SIP v1 (RFC2543), v2(rfc3261) IP/TCP/UDP/RTP/RTCP IP/ICMP/ARP/RARP/SNTP TFTP Client/DHCP Client/ PPPoE Client Telnet/HTTP Server DNS Client NAT/DHCP Server Tone Ring Tone Ring Back Tone Dial Tone Busy Tone Programming Tone Phone Function Codec G.711: 64k bit/s (PCM) G.723.1: 6.3k / 5.3k bit/s G.726: 16k / 24k / 32k / 40k bit/s (ADPCM) G.729A: 8k bit/s (CS-ACELP) G.729B: adds VAD & CNG to G.729 Voice Quality VAD: Voice activity detection CNG: Comfortable noise generator LEC: Line echo canceller Packet Loss Compensation Adaptive Jitter Buffer Call Function Call Hold Call Waiting Call Forward Caller ID 3-way conference Volume Adjustment Speed dial key Phone book Flash IP Assignment Static IP DHCP PPPoE Security HTTP 1.1 basic/digest authentication for Web setup MD5 for SIP authentication (RFC2069/ RFC 2617) QoS ToS field NAT Traversal STUN DTMF Function In-Band DTMF Out-of Band DTMF SIP Info SIP Server Registrar Server (three SIP account) Outbound Proxy Configuration Web Browser Console/Telnet IVR/Keypad Firmware Upgrade TFTP Console HTTP

2 IVR Interface for TA You can use the connected phone to configure the TA. Please follow the instruction to configure your Telephone Adapter. Group IVR Action IVR Menu Choice Parameter(s) Notes: Function unlock keypad #190# None If you want to let user change the setting by keypad. You should hear two beep sounds when unlock is succeed Function lock keypad If you want to lock user change #191# None the setting by keypad. Function Reboot #195# None After you hear Option Successful, hang-up. The system will reboot automatically. (Need Unlock First) Function Factory Reset #198# None System will automatically Reboot. WARNING: ALL User-Changeable NONDEFAULT SETTINGS WILL BE LOST! This will include network and service provider data. (Need Unlock First) Function enable call waiting #138# None Enable Call waiting Function disable call waiting #139# None Disable call waiting Info Check LAN port IP IVR will announce the current IP #120# None Address address of the TA Info Check IP Type #121# None IVR will announce if DHCP in enabled or disabled. Info Check the Phone IVR will announce current in use #122# None Number VoIP number Info Check Network IVR will announce the current Mask #123# None network mask of the TA. Info Check Gateway IP IVR will announce the current Address #124# None gateway IP address of the TA. Info Check Primary IVR will announce the current DNS Server #125# None Setting setting in the Primary DNS field. Info Info Check WAN port IP Address #126# None Check Firmware Version #128# None Setting Set DHCP client #111# None Setting Set Static IP Address #112xxx*xxx*xxx*xxx# Setting Set Network Mask #113xxx*xxx*xxx*xxx# Setting Set Gateway IP Address #114xxx*xxx*xxx*xxx# Enter IP address using numbers on the telephone key pad. Use the * (star) key when entering a decimal point. Enter value using numbers on the telephone key pad. Use the * (star) key when entering a decimal point. Enter IP address using numbers on the telephone key pad. Use the * (star) key when entering a decimal point. IVR will announce the current IP address (WAN Port) of the TA IVR will announce the version of the firmware running on the TA. The system will change to DHCP Client type (Need Unlock First) DHCP will be disabled and system will change to the Static IP type. (Need Unlock First) Must set Static IP first. (Need Unlock First) Must set Static IP first. (Need Unlock First)

Setting Set Primary DNS Server #115xxx*xxx*xxx*xxx# Setting Set Codec #130+[1-8]# Setting Setting Set Handset Gain Set Handset Volume #132+[00~12]# Enter IP address using numbers on the telephone key pad. Use the * (star) key when entering a decimal point. 1:G.711 u-law, 2: G.711 a-law, 3: G.723.1, 4: G.729a, 5: G.726 16K, 6: G.726 24K, 7: G.726 32K, 8: G.726 40K, #131+[00~15]# Handset Gain from 0~15 Handset Volume from 0~12 Must set Static IP first. (Need Unlock First) You can set the codec you want to the first priority. (Need Unlock First) You can set the Handset gain to proper value, default is 6 (Need Unlock First) You can set the Handset volume to proper value, default is 10 (Need Unlock First) 3 Setup the TA by Web Browser The TA provides a built-in web server. You can use Web browser to configure the TA. First please input the IP address in the Web page. In the end of IP address, please add the port number :9999. Ex: http://192.168.123.1:9999 The default LAN port IP is 192.168.123.1, with DHCP Server enabled. The default WAN port has DHCP client enabled; the IP would be automatically assigned if it is connected to a network with DHCP server. 3.1 First Time Login 1. Connected your PC to the LAN port of the Adapter, and your DSL modem or router to the WAN port as the following diagram. 2. Enable the Obtain an IP address automatically in your Local Area Connection inside Windows, under Control Panel -> Network Connections -> Local Area Connection -> Property -> Internet Protocol -> Properties

Your PC would then obtain an IP from the Adapter. 3. Inside your PC, open the browser (Internet Explorer), and enter http://192.168.123.1:9999 as follow to enter the web interface of the Adapter 4. Please input the username and password into the blank field. The default setting is: For normal user, the username is: user; and the password is: test. If you use the account login, but you can not configure the SIP setting. For Administrator, the username is: root; and the password is: test. If you use the account login, you can configure all the setting. Please inquiry your service provider if the password is not correct. Troubleshooting Tips: If there is no login page (shown on the left) appears. Please check: i. The link typed in the web browser (Internet Explorer). Make sure http:// and the port :9999 is entered, such that the link is in the form of http://192.168.123.1:9999 ii. The IP address may be changed due to whatever reason. You may plug in a standard phone to the phone, and check the LAN ip address by enter #120# 5. Click the Login button will move into the TA web based management interface. If you change the setting in the Web Management interface, please do remember to click the Submit button in that page. After you finished the change of the setting, click the Save function in the left side, and click the Save Button. When you finished the setting, please click the Reboot

function in the left side, and click the Reboot button in that page. After the system restart, all the setting can work properly. 3.2 Default Reset from Keypads Pick up the handset of the Phone connected to the Phone Adapter Press keys #190# to unlock the phone. You should hear two beep sounds then. Hook on the handset. Pick up the handset again, and press keys #198# to reset the Adapter to factory default. The Adapter would then reset and reboot. 3.3 Default Setting LAN IP Address: 192.168.123.1 (LAN) WAN IP Address: Dynamic Address/DHCP (WAN) Web Management Page Login Name: root Password: test 4 Application Example 4.1 SIP-to-SIP Calling/Answering Applications: Both parties are registered to SIP server with either fixed real IP or private IP under NAT router. The SIP-to-SIP calling works when both calling and answering parties are registered to SIP server with given registered phone numbers. Configurations: Make sure DHCP Client is ON in the Network / Network settings pages, Select Active ON in the SIP settings / Service Domain pages, Enter the Register Name, Register Password, Proxy Server, and Outbound Proxy, Select ON in NAT settings / STUN setting page, if Outbound Proxy is NOT available. Upon successful SIP registration, the LED indicator of PHONE1 would be lighted. Callings: Pick up the phone, and you should hear a dial tone. If you hear a busy tone, most likely the server registration failed. Please go to SIP settings / Service Domain page to review the settings If you re going to call user 1688, please press 1 + 6 + 8 + 8 + #. Note that # key will let the TA dial out the number immediately. Dialing without # will not dial out until the auto dial timer (default=5 seconds) elapsed. 4.2 SIP to Direct IP Calling Applications: The application is for the calling party with ADSL connection as in either Diagrams A or B. The calling party is registered to SIP server with either fixed real IP or private IP under NAT router. The answering party is with fixed real IP. Configurations: Same as in Example 4.1 Select ON in NAT settings / STUN setting page, if Outbound Proxy is NOT available. Upon successful SIP registration, the LCD will show registered <phone number>.

Callings: Press Hand-Free key for speakerphone, and you should hear a dial tone. Press 211*21*191*4# or 211*21*191*4 to call the party with the real IP address of 211.21.191.4. In a moment, you should hear a ring back tone, and wait for the VoIP called party to answer. 4.3 Direct IP to Direct IP Calling/Answering Applications: The applications are for ADSL connection without NAT router as in Diagram A. Both parties are with fixed real IP. The Direct IP calling works when both calling and answering parties are with known fixed IP. SIP server registrations are not required in this application. Configurations: Select Fixed IP, and bridge ON in the Network / Network settings page Enter the items of IP, Subnet Mask, Gateway IP Click the Submit button. Callings: Pick up the phone, and you should hear a dial tone. Press 211*21*191*4# or 211*21*191*4 to call the party with the real IP address of 211.21.191.4. Note that # key will dial out the number immediately. Dialing without # will not dial out until the auto dial timer (default=5 seconds) elapsed. In a moment, you should hear a ring back tone, and wait for the VoIP called party to answer. 4.4 Direct IP to Direct IP Calling within NAT Router Applications: For the calling party in ADSL connection with NAT router as in Diagram B, this Direct IP calling can work when the answering parties are with fixed private IP addresses within the same VPN network, or with fixed real IP addresses. Configurations: Select Fixed IP, and bridge ON in the Network / Network settings page Enter the items of IP, Subnet Mask, Gateway IP Click the Submit button Callings: Pick up the phone, and you should hear a dial tone Press 192*168*1*51# or 192*168*1*51 to call the party with the private IP address of 192.168.1.51. Press 211*21*191*4 to call the party with the real IP address of 211.21.191.4. In a moment, you should hear a ring back tone, and wait for the called party to answer. 4.5 Calling through the FXO Port (LINE Port) Applications: User can make call through the FXO if the TA has the FXO Port (Line Port on the back),. Configurations: Connect the FXO Port (LINE Port) to the PSTN Landline Callings: Pick up the phone, and press 0 + *, you should then hear a dial tone Press the landline phone number, and the call would be made.

4.6 3-Way Conference Call, Call Waiting, Call Hold 4.6.1 3-Way Conference Calling Application This is for 3-way conference call among Parties A, B, and C. Three parties are registered to SIP server with either fixed real IP or private IP. Make a phone call from Party A to the first phone number Party B After the first call is established, press Flash key from Party A to hold the call, and Party A should hear a dial tone Press #512# + Party C s Phone Number + #. Party C would be connected. After the second call is established, press FLASH key from Party A to join in Party B and Party C together for three-way conference call 4.6.2 Call Waiting Application: When a new call is coming while you are talking, you can push the FLASH key on your phone to switch to the new call. You can push the FLASH key to switch between the two calls. 4.6.3 Call Hold Application: You may push the FLASH key to hold the current call for a while, then push FLASH key again to resume talking. 4.6.4 Call Transfer Blind Transfer You can transfer the current call to another user using the Call Transfer function Party B Call Party A. Party A and Party B are in conversation Party A press FLASH to hold the call. A should hear a dial tone Party A can then press #510# + Party C s Phone Number + #, to make blind transfer to Party C. 4.6.5 Call Transfer Attendant Transfer Party B Call Party A. Party A and Party B are in conversation Party A press FLASH to hold the call. A should hear a dial tone Party A can then press #511# + Party C s Phone Number + #, Party C s phone would ring, and Party A can talk to Party C. When Party A hang up the call, Party B and Party C would in conversation. 4.6.6 Call Forward You can forward the incoming call to a third-party Set up the Call Forward Number using the Web Configuration (Refer to 3.2.1) or Using the LCD Menu ( Phone Setting -> Call Forward ) Press the Key FORWARD to enable the Call Forward 5 Phone Configuration 5.1 System Information When you login the web page, you can see the TA current system information like firmware version, company etc in this page. Also you can see the function lists in the left side. You can use mouse to click the function you want to set up.

5.2 Phone Book In Phone Book contains Speed Dial Settings. You can setup the Speed Dial number. If you want to use Speed Dial you just dial the speed dial number (from 0~9) then press #. Phone: Phone Book Entry Index. The total number of Phone Book Entry is 140 Name: Phone Book Entry Name. Check Example for details URL: Phone Book Entry URL: the lin number or IP. Check example for details

Example Example 1: Name: 301, URL: 301@192.168.1.2 If user dialed [301#], the phone would dial out [192.168.1.2] actually. Example 2: Name: 206, URL: 17476433364 If user dialed [206#], the phone would dial out [17476433364] actually. Example 3: Name: 202, URL: 192.168.1.202:5062 If user dialed [202#], the phone would dial out [192.168.1.2:5064] actually. 5.3 Phone Setting In Phone setting contains Call Forward, SNTP Settings, Volume Settings, Block Setting, Auto Answer, Caller ID, Auto Dial Setting, Flash Time Setting and Call Waiting Setting functions. 5.3.1 Call Forward function User can setup the phone number you want to forward in this page. There are three type of Forward mode. You can choose All Forward, Busy Forward, and No Answer Forward by click the icon.

All Forward: All incoming call will forward to the number you choose. You can input the name and the phone number in URL/Number field. If you select this function, then all the incoming call will direct forward to the speed dial number you choose. Busy Forward: If you are on the phone, the new incoming call will forward to the number you choose. You can input the name and the phone number in URL field. No Answer Forward: If you can not answer the phone, the incoming call will forward to the number you choose. You can input the name and the phone number in URL field. Also you have to set the Time Out time for system to start to forward the call to the number you choose.

5.3.1.1 Call Forward from VoIP to PSTN (Only supported by model with FXO port) 1. Select the case to forward (e.g. All Forward), Click option of PSTN 2. Enter the PSTN phone number in the All Fwd No. -> URL/Number field, and the Name Field 3. Click Submit to take effect. 5.3.1.2 Call Forward from PSTN to VoIP (Only supported by model with FXO port) 1. Select the case to forward (e.g. All Forward), Click option of IP 2. Enter the PSTN phone number in the All Fwd No. -> URL/Number field, and the Name Field 3. Click Submit to take effect. 5.3.2 SNTP setting function User can setup the primary and second SNTP Server IP Address, to get the date/time information. Also you can base on your location to set the Time Zone, and how long need to synchronize again. When you finished the setting, please click the Submit button. 5.3.3 Volume setting function User can setup the Handset Volume, Ringer Volume, and the Handset Gain. When you finished the setting, please click the Submit button. Handset Volume is to set the volume for you can hear from the handset. PSTN-Out Volume is to set the PSTN volume for you can hear. Handset Gain is to set the volume send out to the other side s handset. PSTN-In Gain is to set the volume send out to the other side s handset.

5.3.4 DND function You can setup the Block Setting to keep the phone slience. You can choose Always Block or Block a period. DND Always: All incoming call will be blocked until disable this feature. DND Period: Set a time period and the phone will be blocked during the time period. If the From time is large than the To time, the Block time will from Day 1 to Day 2. When you finished the setting, please click the Submit button. 5.3.5 Auto Answer function (Supported by FXO Model Only) You can set the Auto Answer function to answer the incoming call by the phone. If the call is come from the IP, then the TA can let user to redial the call to PSTN phone number. If the call is coming from PSTN, then the TA can let user to redial to IP Phone number. Auto Answer Counter is to set after the ring counts meet the number you set then the auto answer will enable. Auto Answer: Off Disable Auto Answer IP IN Enable Auto Answer on incoming VoIP Call

FXO IN Enable Auto Answer on incoming FXO (landline) Call Both Enable Auto Answer for both incoming VoIP and FXO (Landline) Calls Trunk Gateway Use ATA as a Trunk Gateway, refer to 3.3.6 for details Auto Answer Counter: The number of ring count before the ATA automatically answer the call PIN Code Enabled: If enabled, auto-answered call will prompt user (a beep-beep sound) to enter a PIN. PIN has to be ended by a # key. (e.g., if the PIN is 123, user has to press 1+2+3+#) If PIN is correct, a second dial tone will be heard to prompt entering the phone number. PIN Code: The code requires the user to enabled if PIN Code is enabled 5.3.6 Trunk Gateway (Supported by FXO Model Only) You can use the ATA with FXO as a Trunk Gateway Let an IP PBX or SIP Proxy to dial to PSTN through the ATA s FXO port. 1. Under Auto Answer page, set the Auto Answer option as Trunk Gateway 2. On an IP PBX, add a Outgoing SIP Trunk. On the SIP Outgoing Trunk, set the Gateway IP to the IP or domain name of the ATA. So, the ATA will dial out incoming SIP call number through the FXO port. Here is a example of setting a Outgoing SIP Trunk with an IP PBX

When User of the IP PBX dial 75698088, the IP PBX will route the call to the ATA (192.168.1.207), the ATA will make an outgoing call to number 5698088 5.3.7 Caller ID function User can set the device to show Caller ID in your PSTN Phone or IP Phone. There are four selection of Caller ID. You need to base on your environment to set the Caller ID function for FSK or DTMF. 5.3.8 Flash Time Setting function When you use the PSTN Phone and you need to press the Hook to do the Flash (Switch to the other phone line or HOLD), this function is for you to set the time you press the Hook to represent the Flash function

5.3.9 Call Waiting setting function You can Enable/Disable the Call Waiting function, when you are talking with someone, there is a new incoming call, you will hear the call waiting tone. 5.3.10 T.38 Fax setting function Our Phone Adapter support T.38 fax function. To use Fax in T.38 mode, 1. Under T.38 Setting Page, select T.38 as On 2. Submit and let the Adapter Reboot T.38 Port Port: The network port that used for T.38 communication. 5.3.11 Dial Plan 5.3.11.1 Dial Now TA would dial the number out immediately if it matches the Dial Now Rule. Dial Now Rule has the same syntax as the Dial Plan above.

Rule Syntax 0-9 Number Digit e.g. 911 Dialed numbers 911 + or Or Separator e.g. 911+18503 Dialed number 911 or 18503 [] Range e.g. 00[2-4] Dialed number 002, 003 or 004 x Any Number e.g. 86xx Any 4 digits dialed number start with 86 5.3.11.2 Add or Replace Prefix You can use the Dial Plan to add prefix to phone number automatically before dialing out. There are 4 replace rules, each the dialed number matches the rule, a prefix would be added (or replace) to the dialed number. Rule Syntax 0-9 Number Digit e.g. 86 Dialed numbers start with 86 + or Or Separator e.g. 002+005 Dialed number start with 002 or 005 [] Range e.g. 00[2-4] Dialed number start with 002, 003 or 004 x Any Number e.g. 86xx Any 4 digits dialed number start with 86 Example 1: Append, Append Prefix: 002, Rule: 8613 8662 1:If what we dialed begins with 8613, number 002 when is added to the beginning; so the actual number dialed would be [002 8613 xxx]. 2:If what we dialed begins with 8662, number 002 when is added to the beginning; so the actual number dialed would be [002 8663 xxx]. Example 2:Replace, Append Prefix: 006, Rule: 00[2-9] 1:If what we dialed begins with 002 (e.g. 0021386), the number 002 would be replaced with 006 ; so the actual number dialed would be like 0061386. 2:If what we dialed begins with 003, the number 003 would be replaced with 006 ; so the actual number dialed would be [006 xxx].

Example 3: Append, Add Prefix: 009, Rule: 12 1:If what we dialed begins with 12, the number 12 would be prefixed by number 009 ; so the actual number dialed would be [009 12 xxx]. Example 4: Append, Add Prefix 007, Rule: 5xxx 35xx 21xx 1:If what we dialed begins with 5, and followed any 3 numbers, the number would be prefixed by number 007 ; so the actual number dialed would be [007 5xx]. 2:If what we dialed 534, the number 5 is only followed by 2 numbers, it does not match with the dialing rule; so the actual number dialed is still [534]. 3:If what we dialed begins with 35, and followed any 2 numbers, the number would be prefixed by number 007 ; so the actual number dialed would be [007 35xx]. 4:If what we dialed 358822, the number 5 is only followed by 2 numbers, so it does not match with the dialing rule; so the actual number dialed is still [358822]. 5.3.11.3 Auto Dial Time TA would auto dialed out the numbers if no more key is pressed for a few seconds Auto Dial Timeout. The Auto Dial Time can be set in this Dial Plan page, the default timeout is 5 seconds 5.3.12 Dial Plan Call Routing (Supported by FXO Model Only) If the dialed number matched the Routing rule, the call would be dial to the route specific in the Routing to field. For example. Routing to: FXO Routing rule: [2-4]xxxxxx 911 [2-4]xxxxxx mean if any dialed number begins with 2 to 4, and followed by 6 digits would be dialed to FXO(PSTN). E..g 3234567 would be dialed through the FXO(PSTN) 911 mean if that if user dialed 911, the number would be routed through FXO port. 5.4 Network In Network you can check the Network status, configure the Network Settings and DDNS settings. 5.4.1 Network Status You can check the current Network setting in this page.

5.4.2 WAN Settings You can configure the TA Network setting in this page. 5.4.2.1 Mode There are two modes for the WAN port: Bridge or NAT. Bridge mode: The two WAN and LAN Ethernet ports will be bridged and transparent. NAT mode: The embedded NAT will be enabled.

The IP type for IP Phone is default at Fixed IP (192.168.1.100). You may select a proper IP type for your network requirements. 5.4.2.2 NAT Settings To enable embedded NAT, you must set mode of WAN port to NAT. This embedded NAT is useful for ADSL users without NAT router, and it separates the WAN port from the LAN port to perform router IP address translation. For the WAN port, please select a proper IP type: DHCP Client(default): Automatically get an assigned IP from the your router. Please select this if there is a router in your network. PPPoE: Most commonly way to connect to your Internet Service Provider through ADSL. Enter the given username and password from the Internet Service Provider. Fixed IP: Assign a Fixed IP to the Adapter. MAC settings for LAN and WAN have been preprogrammed and must be different from each other. Please leave it default unless necessary. After you finished the setting, please click the Submit button. 5.4.3 LAN Settings 5.4.3.1 LAN Setting IP: The IP Address of the LAN Port. Default is 192.168.123.1 5.4.3.2 DHCP Server There is built-in DHCP Server in the Phone Adapter. The default is enabled. Start IP/End IP: The range of IP to be assigned. E.g. Assign IP from 192.168.1.150 to 192.168.1.200 Lease Time: The lease time of each DHCP assignment 5.4.4 DDNS Setting You can configure the DDNS setting in this page. You need to have the DDNS account and input the information properly. You can have a DDNS account with a public IP address then others can call you via the DDNS account. But now most of the VoIP applications are work with a SIP Proxy Server. When you finished the setting, please click the Submit button.

5.4.5 VLAN Setting Our Phone Adapter support VLAN. Note that building a VLAN network requires a switch support VLAN. VLAN can be used to separate VOIP traffic from other DATA traffic. Then, user can configure the VLAN switch to handle the VOIP VLAN tag in higher priority. This is one of the ways to guarantee Quality-of-Service within a private network. 5.4.6 PPTP Setting You can connect the Phone Adapter to a VPN (PPTP) Server. Using VPN, user can connect to a SIP Server (SIP Proxy or IP PBX) inside a private network, or making a secure phone call through

the Adapter. PPTP Server: VPN Server to connect. PPTP Username/Password: Username and Password for the VPN account of the VPN Server 5.4.7 STUN Setting You can setup the STUN Enable/Disable and STUN Server IP address in this page. This function can help your TA working properly behind NAT. To change these settings please following your ISP information. When you finished the setting, please click the Submit button. 5.5 SIP Settings You can setup the Service Domain, Port Settings, Codec Settings, RTP Setting, RPort Setting and Other Settings for SIP Proxy Server registrations in this page. 5.5.1 Service Domain You may register up to three SIP accounts in the IP Phone. You can call your friends via firstly enabled SIP account and receive the phone calls from all the three SIP accounts. It supports 3 services; allow user register on different service providers. Click Active ON to enable the Service Domain, and then input the following fields: Realm (1 ~ 3) Display Name: User Name: Register Name: Register Password: Domain Server: Proxy Server: Outbound Proxy: Server, you may skip this. MWI: Name you want to display. User Name given by your ITSP. Register Name given by your ITSP. Register Password given by your ITSP. Domain Server given by your ITSP. Proxy Server given by your ITSP. Outbound Proxy of ITSP. If not provided or it is the same as Proxy Whether you would like to subscribe for Voicemail notification. When it shows Registered in the Register Status, it indicates a successful registration to the

ITSP, and the PHONE LED will start flashing. The IP Phone is then ready for VoIP call. If you have more than one SIP account, please follow the steps to register to other ITSPs. After you finished the setting, please click the Submit button. Switching Default SIP Account (Domain) If you have registered more than one SIP Accounts, you may want to switch to default SIP account for outgoing call. 1. Please pick up the handset, and Press "1*#" - Make first SIP account as default. Press "2*#" - Make second SIP account as default Press "3*#" - Make third SIP account as default 2. Put down the handset, and the new default account would be used for outgoing call. 5.5.2 Port Settings You can setup the SIP and RTP port number in this page. SIP Port: Network Port for SIP communication. Default is 5060 RTP Port: Network Port for RTP data communication. Default is 60000 Each ISP provider will have different SIP/RTP port setting, please refer to the ISP to setup the port number correctly. When you finished the setting, please click the Submit button. 5.5.3 Codec Settings You can setup the Codec priority, RTP packet length, and VAD function in this page. You need to follow the ISP suggestion to setup these items. When you finished the setting, please click the Submit button.

5.5.4 Codec ID Setting Sometimes different VoIP devices with different Codec ID will cause the interoperability issue. If you are talking with others got some problems, you may ask the other one what kind of Codec ID he use, then you can change your Codec ID. When you finished the setting, please click the Submit button.

5.5.5 DTMF Setting User can setup the InBand DTMF, 2833 Out-Band DTMF and Send DTMF SIP Info Enable/Disable in this page. To change this setting, please follows your ISP information. When you finished the setting, please click the Submit button. DTMF Setting: Configure how the Phone Adapter sends out the DTMF signal. 2833: Send in format compliant to RFC2833 Inband DTMF: Send the DTMF as Inband DTMF Send DTMF SIP Info: Send the DTMF as SIP Info 5.5.6 RPort Function: You can setup the RPort Enable/Disable in this page. To change this setting, please following your ISP information. When you finished the setting, please click the Submit button. 5.5.7 Other Settings You can setup the Hold by RFC, Voice/SIP QoS and SIP expire time in this page. To change these settings please following your ISP information. When you finished the setting, please click the Submit button. The QoS setting is to set the voice packets priority. If you set the value higher than 0, then the voice packets will get the higher priority to the Internet. But the QoS function still need to cooperate with the others Internet devices.

5.6 Advanced 5.6.1 Auto Configuration Setting You can setup the Auto Configuration so that the ATA would automatically download the configure file from server. This function helps server provider for their deployment. ATA 1 (MAC:001122334401) ATA 2 (MAC:001122334402) ATA 3 (MAC:001122334403) Get Configuration Get Configuration Get Configuration Server (TFTP/HTTP/FTP) Configuration Files: 001122334401.dat 001122334402.dat 001122334403.dat

For example, you would like to setup the auto configuration using an FTP Server FTP Server Domain name: ftp.myserver.com FTP Username: user FTP File Path: /file/firmware 1. Setup the configuration files, and put them on your FTP Server, under the File Path. (Please refer to Auto Configuration and StandardCfg.dat document for details) 2. Under the Auto Configuration Setting of ATA: Set the following: 3. The ATA would automatically reboot, and get the updated configuration file from server. 5.6.2 FXS/FXO Setting: You can setup the FXS/FXO setting in this page. When you are using different country s PSTN Phone or connect to different country s PSTN Line, you have to set the country s setting to meet the requirement. When you finished the setting, please click the Submit button.

5.6.3 Advanced Setting ICMP Not Echo: Enable/Disable the ICMP Echo in this page. This function can disable echo when someone ping this device, it can avoid hacker try to attack the device. When you finished the setting, please click the Submit button. Management from WAN: Enable/Disable web management page access from WAN You

port Web Server Port: Configure the Web management server port. Default is 9999 PPPoE Retry period: Configure the PPPoE Retry period Send Anonymous CID: Enable/Disable sending anonymous Caller ID. Anonymous Call Rejection: Enable/Disable Anonymous Call Rejection. Enable this would automatically reject anonymous call. Billing Signal: Configure how the adapter generates the Billing Signal. Switch to PSTN when No SIP registration: Adapter would automatically switch to PSTN/FXO mode if there is no SIP registration Send Flash event: Configure how the adapter send the Flash Event 5.7 System Auth 5.7.1 In System Authority You can change your login name and password.. When you finished the setting, please click the Submit button. 5.8 Save Change 5.8.1 In Save Change You can save the changes you have done. If you want to use new setting in the TA, You have to click the Save button. After you click the Save button, the TA will automatically restart and the 5.9 Update 5.9.1 In Update You can update the TA s firmware to the new one or do the factory reset to let the TA back to default setting. 5.9.2 In New Firmware You can update new firmware via HTTP in this page. You can upgrade the firmware by the following steps: Select the firmware code type, Risc or DSP code. Click the Browse button in the right side of the File Location or you can type the correct path and the filename in File Location blank. Select the correct file you want to download to the TA then click the Update button.

5.9.3 In Default Setting You can restore the TA to factory default in this page. You can just click the Restore button, then the TA will restore to default and automatically restart again. 5.10 Reboot 5.10.1 Reboot You can restart the TA. If you want to restart the TA, you can just click the reboot button, then the TA will automatically.

6 Troubleshooting 6.1 The WAN LED does not light Make sure the WAN port of Phone Adapter is connected properly to your ADSL Modem or Router. Power Reset and check again. 6.2 Do not hear DIAL TONE? No dial tone happens when there is Internet connection problem or when the Service registration failed. Check if the Account information is set up correctly. Goto SIP Settings -> Service Domain of the Web Management Interface. Make sure all the information is entered properly, especially is the password. Press Submit and Reboot the IP Phone Make sure the Phone Adapter can connect to the Internet. Check if your ADSL Modem is working properly. Call your VoIP Service provider, to check if there is any problem with their server. 6.3 How to Reset the Phone Adapter to Factory Default? Refer to Ch. 3.2 6.4 How to switch to another SIP account for outgoing call? If you have registered more than one SIP Accounts, you may want to switch to default SIP account for outgoing call. 1. Please pick up the handset, and Press "1*#" - Make first SIP account as default. Press "2*#" - Make second SIP account as default Press "3*#" - Make third SIP account as default 2. Put down the handset, and the new default account would be used for outgoing call.